RTC_[D]CHECK_op: Remove "u" suffix on integer constants

webrtc/modules/audio_device/android/opensles_player.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 127 matching lines @@
     return 0;
   }
   // Stop playing by setting the play state to SL_PLAYSTATE_STOPPED.
   RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_STOPPED), -1);
   // Clear the buffer queue to flush out any remaining data.
   RETURN_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_), -1);
 #if RTC_DCHECK_IS_ON
   // Verify that the buffer queue is in fact cleared as it should.
   SLAndroidSimpleBufferQueueState buffer_queue_state;
   (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &buffer_queue_state);
-  RTC_DCHECK_EQ(0u, buffer_queue_state.count);
-  RTC_DCHECK_EQ(0u, buffer_queue_state.index);
+  RTC_DCHECK_EQ(0, buffer_queue_state.count);
+  RTC_DCHECK_EQ(0, buffer_queue_state.index);
 #endif
   // The number of lower latency audio players is limited, hence we create the
   // audio player in Start() and destroy it in Stop().
   DestroyAudioPlayer();
   thread_checker_opensles_.DetachFromThread();
   initialized_ = false;
   playing_ = false;
   return 0;
 }
 
@@ skipping 257 matching lines @@
   RTC_DCHECK(player_);
   SLuint32 state;
   SLresult err = (*player_)->GetPlayState(player_, &state);
   if (SL_RESULT_SUCCESS != err) {
     ALOGE("GetPlayState failed: %d", err);
   }
   return state;
 }
 
 }  // namespace webrtc